Power gains and capacity gains for a relaxed frame erasure rate

ABSTRACT

A method of controlling frame transmissions includes determining, at a first device, a frame erasure rate for a communication session between the first device and at least a second device. The method also includes comparing the frame erasure rate to an erasure threshold. The method further includes discarding an active speech frame if the frame erasure rate satisfies the erasure threshold.

I. CROSS REFERENCE TO RELATED APPLICATIONS

The present application claims the benefit of U.S. Provisional PatentApplication No. 62/244,142, entitled “IMPROVED POWER GAINS AND CAPACITYGAINS FOR A RELAXED FRAME ERASURE RATE,” filed Oct. 20, 2015, which isexpressly incorporated by reference herein in its entirety.

II. FIELD

The present disclosure is generally related to frame erasure rates.

III. DESCRIPTION OF RELATED ART

Advances in technology have resulted in smaller and more powerfulcomputing devices. For example, there currently exist a variety ofportable personal computing devices, including wireless telephones suchas mobile and smart phones, tablets, and laptop computers that aresmall, lightweight, and easily carried by users. These devices cancommunicate voice and data packets over wireless networks. Further, manysuch devices incorporate additional functionality such as a digitalstill camera, a digital video camera, a digital recorder, and an audiofile player. Also, such devices can process executable instructions,including software applications, such as a web browser application, thatcan be used to access the Internet. As such, these devices can includesignificant computing capabilities.

A source device may encode active frames (e.g., speech frames) andinactive frames at different bit-rates and send the encoded frames to adestination device. As a non-limiting example, according to the ThirdGeneration Partnership Project (3GPP) Enhanced Voice Services (EVS)standard, active frames may be encoded at 13.2 kilobits per second(kbps) and inactive frames may be encoded at 2.4 kbps. A redundant copy(e.g., a partial copy) of a previous active frame (N−X) may be encodedand combined with a current active frame (N).

By attaching partial copies of previous frames to current frames, thenumber of active frames that are unaccounted for (e.g., the frame“erasure rate”) at the destination device may be relatively low.Different standards may set different frame erasure rates forcommunication sessions (e.g., voice calls). As a non-limiting example,EVS channel aware mode may tolerate up to a ten percent frame erasurerate for a communication session between the source device and thedestination device and still deliver the same voice quality as AdaptiveMulti-Rate Wideband (AMR-WB) subject to two percent frame erasure rate(commercial grade HD voice services). If the frame erasure rate at thedestination device is relatively low and is less than the tolerableframe erasure rate of the EVS channel aware mode frame rate, additionalmargin (in the form of dropped frames) may be introduced to improvecommunications while maintaining the same voice quality as AMR-WB at twopercent frame erasure rate. As described below, the margin may be usedto improve power gains at the source device and capacity gains of asystem that includes the source device and the destination device.

IV. SUMMARY

According to one implementation of the techniques disclosed herein, amethod of controlling frame transmissions includes determining, at afirst device, a frame erasure rate for a communication session betweenthe first device and at least a second device. The method also includescomparing the frame erasure rate to an erasure threshold. The methodfurther includes discarding an active speech frame if the frame erasurerate satisfies the erasure threshold.

According to another implementation of the techniques disclosed herein,an apparatus includes a rate monitor configured to determine a frameerasure rate for a communication session between a first device and atleast a second device. The apparatus also includes comparison circuitryconfigured to compare the frame erasure rate to an erasure threshold.The apparatus further includes active frame discard circuitry configuredto discard an active speech frame if the frame erasure rate satisfiesthe erasure threshold.

According to another implementation of the techniques disclosed herein,a non-transitory computer-readable medium includes instructions forcontrolling transmissions. The instructions, when executed by aprocessor, cause the processor to perform operations includingdetermining, at a first device, a frame erasure rate for a communicationsession between the first device and at least a second device. Theoperations also include comparing the frame erasure rate to an erasurethreshold. The operations further include discarding an active speechframe if the frame erasure rate satisfies the erasure threshold.

According to another implementation of the techniques disclosed herein,an apparatus includes means for determining a frame erasure rate for acommunication session between a first device and at least a seconddevice. The apparatus also includes means for comparing the frameerasure rate to an erasure threshold and means for discarding an activespeech frame if the frame erasure rate satisfies the erasure threshold.

According to another implementation of the techniques disclosed herein,a method of controlling a block error rate for a communication channelincludes determining, at a particular device (e.g., a network device, afirst device, or a second device), that a communication session betweenthe first device and the second device supports an Enhanced VoiceServices (EVS) coder/decoder (CODEC). The first device and the seconddevice communicate via the communication channel. The method alsoincludes increasing a block error rate for the communication channel inresponse to determining that the communication session supports the EVSCODEC. The EVS CODEC may support different modes. As a non-limitingexample, the EVS CODEC may support EVS channel aware mode. As anothernon-limiting example, the EVS CODEC may support 13.2 kbps non-channelaware mode.

V. BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a particular illustrative implementation ofa system that is operable to control frame transmissions;

FIG. 2A is a flowchart of a particular implementation of a process forcontrolling frame transmissions;

FIG. 2B is a flowchart of another particular implementation of a processfor controlling frame transmissions;

FIG. 3A is a flowchart of a method for controlling frame transmissions;

FIG. 3B is a flowchart of another method for controlling frametransmissions;

FIG. 4 is a block diagram of a particular illustrative implementation ofa device that is operable to control frame transmissions; and

FIG. 5 is a block diagram of a base station that is operable to controlframe transmissions.

VI. DETAILED DESCRIPTION

As used herein, the term “device” refers to an electronic device thatmay be used for voice and/or data communication over a wirelesscommunication network. Examples of devices include communicationdevices, cellular phones, personal digital assistants (PDAs), handhelddevices, headsets, wireless modems, laptop computers, personalcomputers, etc. The devices described herein may be compatible with oneor more mobile telecommunication technologies. For example, the devicesdescribed herein may be compatible with third generation (3G) mobiletelecommunication technologies, fourth generation (4G) mobiletelecommunication technologies, and/or fifth generation (5G) mobiletelecommunication technologies. Additionally, or in the alternative, thedevices described herein may be compatible with different standards(e.g., a Long-Term Evolution (LTE) wireless communication standard, aLTE advanced (LTE-A) wireless communication standard, a WorldwideInteroperability for Microwave Access (WiMAX) wireless communicationstandard, an Enhanced Voice Services (EVS) standard, an AdaptiveMulti-Rate Wideband (AMR-WB) standard, an LTE-direct (LDE-D) wirelesscommunication standard, etc.).

The average voice activity per user in a typical two-way conversationmay be relatively low. As an illustrative example, fifty to sixtypercent of the time a user may be speaking, and the remaining time maycorrespond to user silence. Active speech may be encoded at a relativelyhigh bit-rate (e.g., 12.2 kbps for AMR and 13.2 kbps for EVS), andinactive frames or background noise may be encoded using discontinuoustransmissions (DTX) at relatively low bit-rates (e.g., 2 kbps). In afull-duplex two-way conversation, use of DTX may provide power savingsby either transmitting low rate packets (Silence Descriptor or SIDframes) during voice inactivity or skipping transmission for NO_DATA orblank frames. In addition, the active time for a terminal to transmitpackets may be reduced based on the voice activity. If the average voiceactivity at a first terminal is sixty percent, then the transmit powerat the first terminal may be potentially reduced by roughly fortypercent. As used herein, the “transmit power” or “transmission power”may refer to a time-averaged transmission power, a total transmissionpower, or an instantaneous transmission power.

“Link budget” may correspond to the maximum path-loss at which acommunication link between the source device and the destination devicesatisfy a block error rate (BLER) for a target data rate. The BLER maybe expressed as the ratio of the number of erroneous transport blocks tothe total number of received transport blocks at the destination device.The communication link may be sustained with a relatively high path lossin EVS channel aware mode, which in turn, may result in a higher linkbudget.

Additionally, in a Voice-Over Long-Term Evolution (VoLTE) system, asignal-to-noise ratio (SNR) at the destination device may be lowered bylowering the transmit power at the source device. For a fixed maximumtransmit power, coverage and user equipment (UE) power savings may berealized by moving an edge of a cell outwards due to the increased powerheadroom. Alternatively, the source device may maintain a relativelyhigh transmit power and voice quality in poor coverage areas may beimproved.

The packet loss rate may also be increased by not transmitting (e.g.,discarding) the packet at the first device (e.g., the transmittingdevice). In this scenario, the power at which the packets aretransmitted would not be lowered; however, power savings may be achievedat the first device due to the reduced active time and/or due to thereduced number of transmissions. Thus, power savings may be achieved bydropping active speech packets using DTX during periods of activespeech.

In case of congestion and/or bad channel conditions (e.g., greater than2 percent BLER), the network/gateway/receiving terminal may request thetransmitting terminal to transmit redundant packets for decoding in casethe original packets do not arrive (or arrive late) at thenetwork/gateway/receiving terminal. Full packet redundancy transmissionmay increase the effective network bandwidth. While the speech qualitymay be improved due to the availability of redundant packets, theend-to-end latency may increase along with an increase in transmissionpower due to full-redundancy transmissions. Another alternative may beto not change the effective network bandwidth and reduce the encodingbit-rate. For example, the bit-rate for EVS may be reduced from 13.2kbps to 7.2 kbps in order to accommodate one-hundred percent full packetredundancy transmission. However, the reduced bit-rate may have a reducespeech quality.

The EVS channel aware mode operates using a constant bit-rate channel(e.g., 13.2 kbps bit-rate channel). A redundant copy of a previouspacket may be a partial copy of the previous packet, and the redundantcopy of the previous packet may be transmitted for critical speechframes (e.g., frames that may cause a significant quality impact at areceiver if lost). The primary frame bit-rate may be reduced (e.g., to9.6 kbps) to accommodate the redundant copy (e.g., 3.6 kbps) such thatthe combination of the primary and redundant frame fits in the constantbit-rate channel. The bit-rate reduction of the primary frame may beperformed in a source controlled manner (depending on thecharacteristics of the input speech) to ensure reduced impact on theoverall speech quality.

The techniques described herein disclose selectively dropping certainpackets at the encoder of the transmitting terminal. For example, thetransmitting terminal may discard (e.g., not transmit) certain packetsand may introduce frame erasure at the encoder and may send a partialcopy of the un-transmitted packets in a future frame. Using the partialredundant copy of the un-transmitted packets, the receiving terminal mayreconstruct the frame that is not transmitted by the transmittingterminal. The transmitting terminal may selectively transmit packetsbased on source signal characteristics and a target drop rate, asdescribed in further detail with respect to FIG. 2A, or based on target“on” time. Alternatively, the transmitting terminal may not transmitpackets at fixed intervals.

In a two-way conversation or in a multi-party conference, theparticipating terminals (e.g., user devices or user equipment) mayoperate in full-duplex mode and power savings due to selectivetransmission of encoder packets in the presence of partial redundancymay reduce the transmit power. In a device-to-device (D2D) push-to-talk(PTT) group communication, users may communicate one at a time in ahalf-duplex mode. In such a scenario, when partial redundancy issupported, in order to reduce the uplink transmit power, thetransmitting terminal may selectively not transmit a first frame to afirst group of participants, not transmit a second frame to a secondgroup of participants, not transmit a third frame to a third group ofparticipants, etc. Alternatively, the terminal may not transmit atregular intervals when transmitting to all N−1 participants.

The techniques described herein may also be applied at an enodeB/networkside to realize power savings due to reduced “on” time. Capacity gains(in terms of number of users per cell) may also be realized on thenetwork due to a reduced average resource block (RB) consumption.Capacity gains may be realized in a Voice-Over Long-Term Evolution(VoLTE) system via relaxed dynamic scheduler delay requirements (e.g.,enodeB grants may be reduced by a factor of the “on” time reduction)thereby freeing up resources to add more users or for otherapplications. Thus, although the techniques described herein aregenerally performed at a user equipment (UE), it should be appreciatedthat the techniques may also be performed at a network level (e.g., atthe enodeB) to realize downlink capacity gains. Uplink capacity gainsmay be realized if the techniques are performed at the UE.

A channel aware mode of the EVS standard may tolerate up to ten percentframe error rate (FER) and deliver substantially the same voice qualityas AMR-WB at two percent FER. Thus, a transmitting terminal by discard(e.g., “blank”) up to ten percent of active frame packets at theencoder.

The packet drop rate (e.g., the packet drop percentage) or “on” time maybe a function of target capacity gains or of target power reduction. Thetarget capacity gains and/or the target power reduction may bedetermined at the UE or may be determined at the network level andcommunicated to the UE. Upon learning the channel conditions (e.g., in ahigh-FER scenario), the encoder may not drop any packets in order to notintroduce quality artifacts. Alternatively, upon learning the channelFER or effective BLER, the encoder may determine a percentage of packetsto drop (or a target “on” time) and selectively transmit packets suchthat the voice quality is maintained at a particular level, for exampleAMR-WB at 2% frame erasure rate

The techniques described herein may be applicable to multi-unicastuplink transmission in an off-network D2D PTT session to reducetransmission uplink power, unicast two-way or multi-unicast off-networkto reduce transmission uplink power, unicast two-way conversations toincrease network coverage, unicast two-way conversations to increasecapacity efficiency (e.g., the number of users able to participate in acommunication session), or a combination thereof.

Referring to FIG. 1, a system 100 that is operable to control frametransmissions is shown. The system 100 includes a first device 102 incommunication with one or more other devices (e.g., a second device 122)via a network 150. The first device 102 may send data to the seconddevice 122 via the network 150 using a first path 152, and the seconddevice 122 may send data to the first device 102 via the network 150using a second path 154.

The first device 102 may communicate with the network 150 via a firstreverse channel 152 a (e.g., a first reverse link) and a first forwardchannel 154 b (e.g., a first forward link). For example, the firstdevice 102 may transmit data to the network 150 using the first reversechannel 152 a, and the first device 102 may receive data from thenetwork 150 using the first forward channel 154 b. The second device 122may communicate with the network 150 via a second reverse channel 154 a(e.g., a second reverse link) and a second forward channel 152 b (e.g.,a second forward link). For example, the second device 122 may transmitdata to the network 150 using the second reverse channel 154 a, and thesecond device 122 may receive data from the network 150 using the secondforward channel 152 b.

The network 150 may include one or more base stations or access pointsto communicate data between the first device 102 and the second device122. As used herein, data (e.g., packets, frames, offset values,acknowledgements, etc.) communicated via the first path 152 correspondsto data transmitted from the first device 102 to the network 150 via thefirst reverse channel 152 a and received at the second device 122 fromthe network 150 via the second forward channel 152 b. In a similarmanner, data communicated via the second path 154 corresponds to datatransmitted from the second device 122 to the network 150 via the secondreverse channel 154 a and received at the first device 102 from thenetwork 150 via the first forward channel 154 b.

The devices 102, 122 may include fewer or more components thanillustrated in FIG. 1. For example, the devices 102, 122 may include oneor more processors, one or more memory units, or both. In a particularillustrative implementation, the first device 102 and/or the seconddevice 122 may be a smart phone, a cellular phone, a mobilecommunication device, a tablet, a PDA, or a combination thereof. Suchdevices may include a user interface (e.g., a touch screen, voicerecognition capability, or other user interface capabilities).

The first device 102 includes a first speech vocoder 104, a memory 105,a receiver (RX) 106, and a transmitter (TX) 108. The first speechvocoder 104 includes an encoder 110, a de jitter buffer 112, a decoder114, comparison circuitry 116, a rate monitor 118, and active framediscard circuitry 119. The second device 122 includes a second speechvocoder 124, a memory 125, a receiver (RX) 126, and a transmitter (TX)128. The second speech vocoder 124 includes an encoder 130, a de-jitterbuffer 132, a decoder 134, comparison circuitry 136, a rate monitor 138,and active frame discard circuitry 139.

In the example illustrated in FIG. 1, the first device 102 is a“transmitting terminal” and the second device 122 is a “receivingterminal.” For example, the first device 102 may transmit frames thatare received by the second device 122. However, in otherimplementations, each device 102, 122 may concurrently operate as areceiving terminal and a transmitting terminal. For example, the firstdevice 102 may transmit frames to the second device 122 via the firstpath 152 (e.g., transmit frames to the network 150 via the first reversechannel 152 a) and concurrently receive frames from the second device122 via the second path 154 (e.g., receive frames from the network 150via the first forward channel 154 b). Additionally, the second device122 may transmit frames to the first device 102 via the second path 154(e.g., transmit frames to the network 150 via the second reverse channel154 a) and concurrently receive frames from the first device 102 via thefirst path 152 (e.g., receive frames from the network 150 via the secondforward channel 152 b).

The encoder 110 of the first speech vocoder 104 may encode data inframes that are scheduled for transmission to the second device 122during a communication session (e.g., a voice call). For example, theencoder 110 may encode a first frame 160, a second frame 162, an Nthframe 163, etc. According to one implementation, N may be an integervalue that is greater than one. For example, if N is equal toforty-five, the encoder 110 may encode forty-five frames that arescheduled for transmission to the second device 122 during thecommunication session. Each frame 160, 162, 163 may be classified as anactive frame (e.g., an active speech frame) or an inactive frame. Asused herein, an active frame may include or correspond to a frame thatis not determined to be inactive by a voice activity detector. As anon-limiting example, an active frame may include a frame that includeshigher audio levels associated with speech as compared to audio levelsassociated with background noise. According to one implementation,active frames may be encoded at 13.2 kbps and inactive frames may beencoded at 2 kbps. According to another implementation, active framesmay be encoded at a bit-rate within a range from 2.8 kbps to 128 kbps.As used herein, an inactive frame may include or correspond to a framethat is determined to be inactive by a voice activity detector. As anon-limiting example, an inactive frame may include a frame thatincludes higher audio levels associated with background noise ascompared to audio levels associated with speech.

Certain frames may include redundancy information (e.g., a partial copy)of previous frames. For example, a redundant copy of a previous activeframe may be encoded and combined with a current active frame. Toillustrate, the first frame 160 and the Nth frame 163 may be activeframes, the first frame 160 may be encoded at a first time, and the Nthframe 163 may be encoded at a second time that is after the first time.The Nth frame 163 (e.g., the current frame) may include a partial copyof the first frame 160 (e.g., a previous frame) such that if the firstframe 160 is lost during transmission, the decoder 134 of the seconddevice 122 may fetch the Nth frame 163 from the de jitter buffer 132 anddecode the partial copy of the first frame 160 to recover speechassociated with the first frame 160. According to one implementation,the current frame (e.g., the primary frame) bit-rate may be reduced to9.6 kbps and the redundant copy of the previous frame may be encoded at3.6 kbps such that the total bit-rate is equal to 13.2 kbps.

The first device 102 may be configured to determine an erasure threshold(e.g., an allowable frame erasure rate) for the communication sessionbetween the first device 102 and the second device 122. As used herein,the “erasure threshold” corresponds to a rate of dropped active frames(or lost active frames) to be satisfied in order to maintain thecommunication session or in order to maintain the communication sessionat a relatively high quality. As a non-limiting example, if the erasurethreshold is equal to ten percent, nine out of every ten active framesencoded by the encoder 110 of the first device 102 may be accounted forby the decoder 134 at the second device 122 in order to maintain thecommunication session or in order to maintain the communication sessionat a relatively high quality.

According to some implementations, the erasure threshold may bedetermined at the network level (e.g., determined at an enodeB (notshown)) and may be communicated to the first device 102 via the network150. According to other implementations, the erasure threshold may bespecified by a standard. For example, the channel aware mode of the EVSstandard may tolerate up to a ten percent frame erasure rate in order tomaintain a communication session between the first device 102 and thesecond device 122 or in order to maintain the communication session at arelatively high quality for example the same voice quality as AMR-WB at2% FER. As another example, the AMR-WB standard may tolerate up to a twopercent frame erasure rate in order to maintain a communication sessionbetween the first device 102 and the second device 122 or in order tomaintain the communication session at a relatively high quality. Thememory 105 may store erasure rate data 107 that indicates the allowableframe erasure rate. The first speech vocoder 104 may retrieve theerasure rate data 107 from the memory 105 to determine the erasurethreshold. For purposes of the following description, the erasurethreshold is assumed to be equal to ten percent in compliance with theEVS channel aware mode. However, it should be understood ten percent ismerely an illustrative non-limiting example and should in no ways beviewed as limiting.

The first device 102 may also be configured to determine a frame erasurerate for the communication session. The frame erasure rate may also bereferred to as the “Real-time Transport Protocol (RTP) loss rate”. TheRTP loss rate may be the percentage of lost RTP packets during thecommunication session. To illustrate, the rate monitor 138 of the seconddevice 122 may monitor the rate of unaccounted for active frames at thesecond device 122. The rate monitor 138 may determine the rate at whichactive frames are lost or dropped during transmission (e.g., determinethe “frame erasure rate”). Channel conditions may affect the frameerasure rate. For example, relatively bad channel conditions (e.g., lowthroughput, channel congestion, etc.) may increase the frame erasurerate. According to some implementations, the frame erasure rate maycorrespond to an average number of unaccounted for active frames over atime period or may correspond to an average number of active frames thatdid not reach the receiver over a time period. The time period may berelatively short (e.g., less one second) to determine an “instantaneous”frame erasure rate. The time period may be longer (e.g., greater thanone second) to determine an “average” frame erasure rate.

As described above, a current frame may include redundancy information(e.g., a partial copy) of a previous frame. Thus, if a previous frame islost during transmission, the decoder 134 may reconstruct the previousframe using the redundancy information of the current frame. Usingredundancy information to reconstruct frames lost during transmissionmay reduce the effective frame error rate. For example, if the firstframe 160 is lost during transmission and the Nth frame 163 includes apartial copy of the first frame 160, the decoder 134 may reconstruct thefirst frame 160 using the Nth frame 163, and the first frame 160 may notbe considered “lost” for purposes related to the effective frame errorrate. After determining the frame erasure rate (e.g., the RTP loss rate)at the rate monitor 138, the transmitter 128 of the second device 122may transmit rate data 170 indicative of the frame erasure rate (e.g.,the RTP loss rate) to the receiver 106 of the first device 102.

The first device 102 may further be configured to compare the frameerasure rate to the erasure threshold. For example, the comparisoncircuitry 116 may compare the frame erasure rate received from thesecond device 122 to the erasure threshold indicated by the erasure ratedata 107. Based on the comparison, the first speech vocoder 104 maydetermine whether to discard an active speech frame that would otherwisebe transmitted to the second device 122. For example, if the frameerasure rate is greater than the erasure threshold, the first speechvocoder 104 may transmit additional active speech frames (or additionalredundancy information) to reduce the frame erasure rate. According tocertain implementations, the communication session may end (based onfactors such as poor signal quality) or have a reduced quality if theframe erasure rate is greater than the erasure threshold. However, ifthe frame erasure rate is lower than the erasure threshold, the activeframe discard circuitry 119 may discard an active speech frame thatwould otherwise be transmitted to the second device 122. Discarding anactive speech frame (e.g., not transmitting the active speech frame) mayconserve transmission power at the first device 102. The active framediscard circuitry 119 (or other circuitry within the speech vocoder 104)may determine which active frame(s) to drop based on techniquesdescribed with respect to FIG. 2A.

The system 100 of FIG. 1 may enable the first device 102 to conservebattery power by discarding active speech frames when the frame erasurerate is lower than the erasure threshold. For example, by discardingactive speech frames (instead of transmitting the active speech frames)when the frame erasure rate is lower than the erasure threshold, batterydrainage associated with the transmitter 108 may be reduced while thecommunication session between the first device 102 and the second device122. Discarding active speech frames may also increase network capacity(e.g., a number of users per cell site).

Referring to FIG. 2A, a flowchart of a particular implementation of aprocess 200 for controlling frame transmissions is shown. The process200 may be performed by the components of the first speech vocoder 104of FIG. 1, the components of the second speech vocoder 124 of FIG. 1, ora combination thereof. As described herein, the process 200 mayrepresent logic used by the first speech vocoder 104 to selectparticular frames to transmit to the second device 122.

At 202, the first speech vocoder 104 may initialize a frame count to oneand may set a target reduction count over M frames (T_(M)) equal to arequired reduction count (TM_(REQ)). The required reduction count(TM_(REQ)) may correspond to the number of frames to be dropped in orderto achieve the target reduction (T). At 204, the first speech vocoder104 may determine whether the frame count is greater than M, where M isa number of frames in a particular transmission window. According to oneimplementation, M may be randomly determined at the speech vocoder 104.If the frame count is greater than M, at 206, the first speech vocoder104 may set an adjustment value (ADJ) equal to the difference betweenthe required reduction count (TM_(REQ)) and an actual reduction countachieved over M frames (A_(M)) (e.g., ADJ=TM_(REQ)−A_(M)). After settingthe adjustment value, the first speech vocoder 104 may modify therequired reduction count (TM_(REQ)) to correspond to the sum of thetarget reduction count over M frames (T_(M)) and the adjustment value(ADJ) (e.g., TM_(REQ)=T_(M)+ADJ). The first speech vocoder 104 may alsoreset the frame count to one.

If the frame count is not greater than M, at 206, the first speechvocoder 104 may determine whether the required reduction count(TM_(REQ)) is equal to zero, at 208. If the required reduction count(TM_(REQ)) is equal to zero, at 208, the first speech vocoder 104 mayinitiate transmission of a current frame (N), at 210. For example, theencoder 110 may encode data in the current frame (N), and thetransmitter 108 may transmit the current frame (N) to the second device122 via the first path 152. If the required reduction count (TM_(REQ))is not equal to zero, at 208, the first speech vocoder 104 may determinewhether a partial copy of a previous frame (N−3) is included with (e.g.,attached to) the current frame (N), at 212. According to the processdiagram 200, a forward error correction (FEC) offset may be equal tothree such that a current frame may include redundancy information of apreviously sent frame having a sequence number that is offset from asequence number of the current frame by three. The first speech vocoder104 may determine whether a partial copy of the previous frame (N−3) isincluded with the current frame (N) based on the FEC offset. Forexample, if the offset between the previous frame (N−3) and the currentframe (N) is equal to the FEC offset, then the first speech vocoder 104may determine that the partial copy of the previous frame (N−3) isincluded with the current frame (N). It should be understood that theFEC offset depicted in the process 200 is merely an illustrativenon-limiting example and should not be construed as limiting.

If the partial copy of the previous frame (N−3) is included with thecurrent frame (N), at 212, the first speech vocoder 104 may determinewhether a full copy of the previous frame (N−3) was sent to the seconddevice 122, at 214. For example, the first speech vocoder 104 maydetermine that the full copy of the previous frame (N−3) was sent to thesecond device 122 if the first speech vocoder 104 did not issue acommand to discard the previous frame (N−3). If a full copy of theprevious frame (N−3) was sent to the second device 122, at 214, thefirst speech vocoder 104 may compare a criticality value (N_(CRIT)) ofthe current frame (N) to a first threshold (CT₁) and may compare acriticality value ((N−3)_(CRIT)) of the previous frame (N−3) to a secondthreshold (CT₂), at 216. As used herein, a “criticality value” indicatesa level of speech in a particular frame. For example, if a frame has arelatively high criticality value, the frame may include a lot of speechand may be of relatively high importance. If a frame has a lowcriticality value, the frame may include a lot of background noise andmay be of relatively little importance. If the criticality value(N_(CRIT)) is less than the first threshold (CT₁) and the criticalityvalue ((N−3)_(CRIT)) is less than the second threshold (CT₂), the firstspeech vocoder 104 may discard the current frame (N), at 224. If thecriticality value (N_(CRIT)) is not less than the first threshold (CT₁)or the criticality value ((N−3)_(CRIT)) is not less than the secondthreshold (CT₂), the first speech vocoder 104 may initiate transmissionof the current frame (N), at 220, and may increment the thresholds (CT₁,CT₂) to reduce the number of frames transmitted. For example, the higherthe thresholds (CT₁, CT₂), the fewer frames that are transmitted to thesecond device 122. The thresholds (CT₁, CT₂) may be initialized andadjusted by the first speech vocoder 104. Initialization of thethresholds (CT₁, CT₂) may be performed based on characteristics of anaudio source. For example, clean speech may be initialized to a firstset of thresholds, noisy speech may be initialized to a second set ofthresholds, music may be initialized to a third set of thresholds, etc.After the transmitting the current frame (N), the first speech vocoder104 may increment the frame count by one, at 222.

If a full copy of the previous frame (N−3) was not sent to the seconddevice 122, at 214, the first speech vocoder 104 may initiatetransmission of the current frame (N), at 220, and may increment thethresholds (CT₁, CT₂). After the transmitting the current frame (N), thefirst speech vocoder 104 may increment the frame count by one, at 222.If the partial copy of the previous frame (N−3) is not included with thecurrent frame (N), at 212, the first speech vocoder 104 may compare thecriticality value (N_(CRIT)) of the current frame (N) to a thirdthreshold (CT₃), at 218. If the criticality value (N_(CRIT)) is not lessthan the third threshold (CT₃), the first speech vocoder 104 mayinitiate transmission of the current frame (N), at 220, and mayincrement the thresholds (CT₁, CT₂, CT₃) to reduce the number of framestransmitted. If the criticality value (N_(CRIT)) is less than the thirdthreshold (CT₃), the first speech vocoder 104 may discard the currentframe (N), at 224.

Thus, the process 200 of FIG. 2A may enable the first speech vocoder 104to select active frames to discard when the frame erasure rate is lowerthan the erasure threshold. According to the process 200, a relativelylarge number of frames may be discarded if the thresholds (CT₁, CT₂,CT₃) are relatively high. According to one implementation, the first andsecond threshold (CT₁, CT₂) may be lower than the third threshold (CT₃).The third threshold (CT₃) may be incremented more often than the firstand second thresholds (CT₁, CT₂) to discard more number of frames wherethe partial copy of the previous frame (N−3) is not included with thecurrent frame (N).

Referring to FIG. 2B, a flowchart of a particular implementation of aprocess 250 for controlling frame transmissions is shown. The process250 may be performed by the components of the first speech vocoder 104of FIG. 1, the components of the second speech vocoder 124 of FIG. 1, ora combination thereof. As described herein, the process 250 mayrepresent logic used by the first speech vocoder 104 to selectparticular frames to transmit to the second device 122. The process 250may be implemented in conjunction with the process 200 of FIG. 2A.

At 252, the first speech vocoder 104 may determine a target discard rate(TDR) over the M frames. The target discard rate (TDR) may correspond tothe number of frames that may be discarded for every M frames. As anon-limiting example, if the target discard rate (TDR) is equal to tenpercent and M is equal to one hundred, then ten frames out of the onehundred frames may be discarded by the first speech vocoder 104. Asdescribed below, the process 250 may provide mechanisms to spread theten frames out over the course of the one hundred frames by selectivelydecreasing the thresholds (CT₁, CT₂, CT₃) described with respect to FIG.2A.

After determining the target discard rate (TDR), the first speechvocoder 104 may set (or reset) an actual discard rate (ADR) to zero andmay set the frame count to one, at 254. Additionally, the first speechvocoder 104 may set (or reset) the thresholds (CT₁, CT₂, CT₃) to theinitial values, as described with respect to the process 200 of FIG. 2A.If the frame count is greater than M, at 256, the first speech vocoder104 may reset the actual discard rate (ADR) back to zero and may resetthe frame count to back to one, at 254. Alternatively, if the framecount is not greater than M, at 256, the first speech vocoder 104 maytransmit or discard the current frame (N), at 258, based on thetechniques described with respect to the process 200 of FIG. 2.

At 260, the first speech vocoder 104 may update the actual discard rate(ADR) for the frame count. As a non-limiting example, if two frames havebeen discarded and the current frame count is ten (e.g., count=10), thenthe first speech vocoder 104 may update the actual discard rate (ADR) totwenty percent. At 262, the first speech vocoder 104 may determinewhether the actual discard rate (ADR) is within a particular percentage(X) of the target discard rate (TDR). As a non-limiting example, thetarget discard rate (TDR) may be ten percent and the particularpercentage (X) may be five percent. In this example, if the actualdiscard rate (ADR) is between five percent and fifteen percent, then thefirst speech vocoder 104 may determine that the actual discard rate(ADR) is within the particular percentage (X) of the target discard rate(TDR).

If the actual discard rate (ADR) is not within the particular percentageof the target discard rate (TDR), at 262, then the first speech vocoder104 may decrease the thresholds (CT₁, CT₂, CT₃), at 266. As describedwith respect to FIG. 2A, decreasing the thresholds (CT₁, CT₂, CT₃) mayresult in the first speech vocoder 104 discarding fewer frames. Thus,decreasing the thresholds (CT₁, CT₂, CT₃) may result in a spreading thediscarded frames over the M frames as opposed to discarding a relativelyhigh number of frames at first and discarding a relatively low number offrames as the frame count approaches M. After decreasing the thresholds(CT₁, CT₂, CT₃), at 266, the first speech vocoder 104 may increment theframe count by one, at 264. If the actual discard rate (ADR) is withinthe particular percentage of the target discard rate (TDR), at 262, thenthe first speech vocoder 104 may bypass adjusting the thresholds (CT₁,CT₂, CT₃) and may increment the frame count by one, at 264.

The process 250 of FIG. 2B may implement techniques to spread thediscarded frames over the M frames as opposed to discarding a relativelyhigh number of frames at first and discarding a relatively low number offrames as the frame count approaches M. Distributing the discardedframes over the M frames preserve audio quality by reducing scenarioswhere the frame dropping is clustered. For example if the target discardrate (TDR) is ten percent and M is one hundred, the techniques of FIG.2B may provide mechanisms such that the ten frames to be discard arespread out and are not all discarded consecutively or within a smalltime window.

Referring to FIG. 3A, a particular implementation of a method 300 forcontrolling frame transmissions is shown. The method 300 may beperformed by the components within the first device 102 of FIG. 1, thesecond device 122 of FIG. 1, or any combination thereof.

The method 300 includes determining, at a first device, an allowableframe erasure rate (e.g., an erasure threshold) for a communicationsession between the first device and a second device, at 302. Forexample, referring to FIG. 1, the first device 102 may to determine theerasure threshold for the communication session between the first device102 and the second device 122. The erasure threshold may be specified bya standard. For example, the channel aware mode of the EVS standard maytolerate up to a ten percent frame erasure rate to maintain acommunication session between the first device 102 and the second device122. As another example, the AMR-WB standard may tolerate up to a twopercent frame erasure rate to maintain a communication session betweenthe first device 102 and the second device 122. The memory 105 may storeerasure rate data 107 that indicates the erasure threshold. The firstspeech vocoder 104 may retrieve the erasure rate data 107 from thememory 105 to determine the erasure threshold.

A frame erasure rate for the communication session may be determined, at304. For example, referring to FIG. 1, the first device 102 maydetermine the frame erasure rate for the communication session. Forexample, the rate monitor 138 of the second device 122 may monitor therate of unaccounted for active frames at the second device 122. Forexample, the rate monitor 138 may determine the rate at which activeframes are lost or dropped during transmission. Upon determining theframe erasure rate at the rate monitor 138, the transmitter 128 of thesecond device 122 may transmit the rate data 170 indicative of the frameerasure rate to the receiver 106 of the first device 102.

The frame erasure rate may be compared to the allowable frame erasurerate, at 306. For example, referring to FIG. 1, the first device 102 maycompare the frame erasure rate to the erasure threshold. For example,the comparison circuitry 116 may compare the frame erasure rate receivedfrom the second device 122 to the erasure threshold indicated by theerasure rate data 107.

The method 300 may further include determining whether to discard anactive speech frame scheduled for transmission to the second devicebased on the comparison, at 308. For example, referring to FIG. 1, basedon the comparison, the first speech vocoder 104 may determine whether todiscard an active speech frame scheduled for transmission to the seconddevice 122. For example, if the frame erasure rate is not lower than theerasure threshold, the first speech vocoder 104 may transmit additionalactive speech frames (or additional redundancy information) to reducethe frame erasure rate. According to certain implementations, thecommunication session may end if the frame erasure rate is not lowerthan the erasure threshold. However, if the frame erasure rate is lowerthan the erasure threshold, the first speech vocoder 104 may discard anactive speech frame scheduled for transmission to the second device 122.Discarding an active speech frame (e.g., not transmitting the activespeech frame) may conserve transmission power at the first device 102.

According to one implementation, the method 300 may include determiningwhether to discard a particular active speech frame if the frame erasurerate is lower than the erasure threshold. Determining whether to discardthe particular active speech frame may include determining if theparticular active speech frame includes a partial copy of a previousframe.

In response to determining that the particular active speech frameincludes the partial copy of the previous frame, the method 300 mayinclude determining whether the previous frame was transmitted to thesecond device. If the previous frame was not transmitted to the seconddevice, the method 300 may include transmitting the particular activespeech frame to the second device. If the previous frame was transmittedto the second device, the method 300 may include comparing a firstcriticality value of the particular active speech frame to a firstthreshold and comparing a second criticality value of the previous frameto a second threshold. If the first criticality value is not less thanthe first threshold or if the second criticality value is not less thanthe second threshold, the method 300 may include transmitting theparticular active speech frame to the second device. If the firstcriticality value is less than the first threshold and the secondcriticality value is less than the second threshold, the method 300 mayinclude discarding the particular active speech frame.

In response to determining that the particular active speech frame doesnot include the partial copy of the previous frame, the method 300 mayinclude determining whether the first criticality value of theparticular active speech frame is less than a third threshold. Themethod 300 may include transmitting the particular active speech frameto the second device if the first criticality value is not less than thethird threshold. The method 300 may also include discarding theparticular active speech frame if the first criticality value is lessthan the third threshold.

Referring to FIG. 3B, a particular implementation of a method 350 forcontrolling frame transmissions is shown. The method 350 may beperformed by the components within the first device 102 of FIG. 1, thesecond device 122 of FIG. 1, or any combination thereof.

The method 350 includes determining, at a first device, a frame erasurerate for a communication session between the first device and at least asecond device, at 352. For example, referring to FIG. 1, the firstdevice 102 may determine the frame erasure rate for the communicationsession between the first device 102 and the second device 122. In otherimplementations, additional devices may be included in the communicationsession. As a non-limiting example, a third device may also be includedin the communication session. The first device 102 may be a user-enddevice, an enodeB device, or another other communication device.

The method 350 also includes comparing the frame erasure rate to anerasure threshold, at 354. For example, referring to FIG. 1, thecomparison circuitry 116 may compare the frame erasure rate receivedfrom the second device 122 to the erasure threshold indicated by theerasure rate data 107. According to one implementation, the method 350may include receiving a signal indicating the erasure threshold prior tocomparing the frame erasure rate to the erasure threshold. For example,a network device (e.g., an enodeB) may determine erasure threshold andsend the signal (indicating the erasure threshold) to the first device102. According to another implementation, the first device 102 may storedata (e.g., the erasure rate data 107) indicating the erasure threshold.According to another implementation, the first device 102 may determinethe erasure threshold based on an algorithm. According to anotherimplementation, the method 350 may include receiving a signal indicatingthe frame erasure rate. The signal may be received from the seconddevice 122. For example, the second device 122 may determine the frameerasure rate for the communication session between the devices 102, 122and send a signal (indicating the frame erasure rate) to the firstdevice 102.

The method 350 also includes discarding an active speech frame if theframe erasure rate satisfies the erasure threshold, at 356. According toone implementation, the frame erasure rate satisfies the erasurethreshold if the frame erasure rate is lower than the erasure threshold.For example, referring to FIG. 1, if the frame erasure rate satisfiesthe erasure threshold, the first speech vocoder 104 may discard anactive speech frame that would otherwise be transmitted to the seconddevice 122. For example, the active frame discard circuitry 119 mayremove the active speech frame from a queue of speech frames to betransmitted to the second device 112. After removing the active speechframe from the queue, the active frame discard circuitry 119 may erase(e.g., delete) the active speech frame. Thus, discarding the activespeech frame includes bypassing transmission of the active speech frameto reduce transmission power at the first device 102. According to oneimplementation, a block error rate for the communication session may bereduced to increase power savings at the first device 102 and to improvelink budget gains.

The methods 300, 350 of FIGS. 3A-3B may enable the first device 102 toconserve battery power by discarding active speech frames when the frameerasure rate is lower than the erasure threshold. For example, bydiscarding active speech frames (instead of transmitting the activespeech frames) when the frame erasure rate is lower than the erasurethreshold, battery drainage associated with the transmitter 108 may bereduced while the communication session between the first device 102 andthe second device 122.

The methods 300, 350 of FIGS. 3A-3B may be implemented by afield-programmable gate array (FPGA) device, an application-specificintegrated circuit (ASIC), a processing unit such as a centralprocessing unit (CPU), a digital signal processor (DSP), a controller,another hardware device, a firmware device, or any combination thereof.As an example, the methods 300, 350 of FIGS. 3A-3B may be performed by aprocessor that executes instructions, as described with respect to FIG.4.

According to one implementation of the present disclosure, a particularmethod or controlling a block error rate for a communication channelincludes determining, at a network device, that a communication sessionbetween a first device and a second device supports EVS channel awaremode. The first device and the second device may communicate via acommunication channel. For example, referring to FIG. 1, a networkdevice (e.g., the first device 102, the second device 122, or anothernetwork device) may determine that the communication session between thedevices 102, 122 supports EVS channel aware mode.

The particular method may also include increasing a block error rate forthe communication channel in response to determining that thecommunication session supports an EVS channel CODEC. For example,referring to FIG. 1, a network device (e.g., the first device, 102, thesecond device 122, or another device) may increase the block error ratefor the communication channel in response to determining that thecommunication session between the devices 102, 122 supports the EVSchannel aware CODEC. The EVS CODEC may support different modes. As anon-limiting example, the EVS CODEC may support EVS channel aware mode.As another non-limiting example, the EVS CODEC may support 13.2 kbpsnon-channel aware mode. The particular method may improve (e.g.,increase) link budget gains/coverage for the communication session byincreasing the block error rate.

Referring to FIG. 4, a block diagram of a particular illustrativeimplementation of a device 400 (e.g., a wireless communication device)that is operable to control frame transmissions is shown. In variousimplementations, the device 400 may have more or fewer components thanillustrated in FIG. 4. In an illustrative implementation, the device 400may correspond to the first device 102 of FIG. 1 or the second device122 of FIG. 1. In an illustrative implementation, the device 400 mayoperate according to the process 200 of FIG. 2A, the process 250 of FIG.2B, the methods 300, 350 of FIGS. 3A-3B, or a combination thereof.

In a particular implementation, the device 400 includes a processor 406(e.g., a CPU). The device 400 may include one or more additionalprocessors 410 (e.g., one or more DSPs). The processors 410 may includethe first speech vocoder 104. In an alternate implementation, the firstspeech vocoder 104 may be included in a different type of processor,such as a CPU (e.g., the processor 406).

The device 400 may include a memory 452 and a CODEC 434. The memory 452may include instructions 456 that are executable by the processor(s)410. The device 400 may include a wireless controller 440 coupled, via atransceiver 450, to an antenna 442. In a particular implementation, thetransceiver 450 may include the receiver 106, the transmitter 108, orboth, of FIG. 1.

The device 400 may include a display 428 coupled to a display controller426. The device 400 may also include a microphone 446 and a speaker 448coupled to the CODEC 434. The CODEC 434 may include a digital-to-analogconverter 402 and an analog-to-digital converter 404. In a particularimplementation, the CODEC 434 may receive analog signals from themicrophone 446, convert the analog signals to digital signals using theanalog-to-digital converter 404, and provide the digital signals to thefirst speech vocoder 104. The first speech vocoder 104 may process thedigital signals. In a particular implementation, the first speechvocoder 104 may provide digital signals to the CODEC 434. The CODEC 434may convert the digital signals to analog signals using thedigital-to-analog converter 402 and may provide the analog signals tothe speaker 448. In a particular implementation, the CODEC 434represents an analog front-end for audio processing that performsfunctions such as gain control and parameter adjustment.

The first speech vocoder 104 may be used to implement hardware inconjunction with the techniques as described herein. Alternatively, orin addition, a software (or combined software/hardware) may beimplemented in conjunction with the techniques described herein. Forexample, the memory 452 may include instructions 456 executable by theprocessors 410 or other processing unit of the device 400 (e.g., theprocessor 406, the CODEC 434, or both) to perform the process 200 ofFIG. 2A, the process 250 of FIG. 2B, the methods 300, 350 of FIGS.3A-3B, or a combination thereof.

In a particular implementation, the device 400 may be included in asystem-in-package or system-on-chip device 422. In a particularimplementation, the memory 452, the processor 406, the processors 410,the display controller 426, the CODEC 434, and the wireless controller440 are included in a system-in-package or system-on-chip device 422. Ina particular implementation, an input device 430 and a power supply 444are coupled to the system-on-chip device 422. Moreover, in a particularimplementation, as illustrated in FIG. 4, the display 428, the inputdevice 430, the speaker 448, the microphone 446, the antenna 442, andthe power supply 444 are external to the system-on-chip device 422. In aparticular implementation, each of the display 428, the input device430, the speaker 448, the microphone 446, the antenna 442, and the powersupply 444 may be coupled to a component of the system-on-chip device422, such as an interface or a controller. The device 400 may include amobile communication device, a smart phone, a cellular phone, a tablet,a PDA, or any combination thereof.

In conjunction with the described implementations, an apparatus includesmeans for determining a frame erasure rate for a communication sessionbetween a first device and at least a second device. For example, themeans for determining the frame erasure rate may include the firstspeech vocoder 104 of FIGS. 1 and 4, the memory 105 of FIG. 1, thesecond speech vocoder 124 of FIG. 1, the processor 406 of FIG. 4, theprocessors 410 of FIG. 4, the CODEC 434 of FIG. 4, or any combinationthereof.

The apparatus may also include means for comparing the frame erasurerate to an erasure threshold. For example, the means for comparing theframe erasure rate to the erasure threshold may include the first speechvocoder 104 of FIGS. 1 and 4, the comparison circuitry 116 of FIG. 1,the second speech vocoder 124 of FIG. 1, the comparison circuitry 136 ofFIG. 1, the processor 406 of FIG. 4, the processors 410 of FIG. 4, theCODEC 434 of FIG. 4, or any combination thereof.

The apparatus may further include means for discarding an active speechframe if the frame erasure rate satisfies the erasure threshold. Forexample, the means for discarding the active speech frame may includethe first speech vocoder 104 of FIGS. 1 and 4, the active frame discardcircuitry 119 of FIG. 1, the second speech vocoder 124 of FIG. 1, theprocessor 406 of FIG. 4, the processors 410 of FIG. 4, the CODEC 434 ofFIG. 4, or any combination thereof.

Referring to FIG. 5, a block diagram of a particular illustrativeexample of a base station 500 is depicted. In various implementations,the base station 500 may have more components or fewer components thanillustrated in FIG. 5. In an illustrative example, the base station 500may include the system 100 of FIG. 1. In an illustrative example, thebase station 500 may operate according to the method 300 of FIG. 3A, themethod 350 of FIG. 3B, or a combination thereof.

The base station 500 may be part of a wireless communication system. Thewireless communication system may include multiple base stations andmultiple wireless devices. The wireless communication system may be aLong Term Evolution (LTE) system, a Code Division Multiple Access (CDMA)system, a Global System for Mobile Communications (GSM) system, awireless local area network (WLAN) system, or some other wirelesssystem. A CDMA system may implement Wideband CDMA (WCDMA), CDMA 1×,Evolution-Data Optimized (EVDO), Time Division Synchronous CDMA(TD-SCDMA), or some other version of CDMA.

The wireless devices may also be referred to as user equipment (UE), amobile station, a terminal, an access terminal, a subscriber unit, astation, etc. The wireless devices may include a cellular phone, asmartphone, a tablet, a wireless modem, a personal digital assistant(PDA), a handheld device, a laptop computer, a smartbook, a netbook, atablet, a cordless phone, a wireless local loop (WLL) station, aBluetooth device, etc. The wireless devices may include or correspond tothe device 400 of FIG. 4.

Various functions may be performed by one or more components of the basestation 500 (and/or in other components not shown), such as sending andreceiving messages and data (e.g., audio data). In a particular example,the base station 500 includes a processor 506 (e.g., a CPU). The basestation 500 may include a transcoder 510. The transcoder 510 may includean audio 508 CODEC. For example, the transcoder 510 may include one ormore components (e.g., circuitry) configured to perform operations ofthe audio CODEC 508. As another example, the transcoder 510 may beconfigured to execute one or more computer-readable instructions toperform the operations of the audio CODEC 508. Although the audio CODEC508 is illustrated as a component of the transcoder 510, in otherexamples one or more components of the audio CODEC 508 may be includedin the processor 506, another processing component, or a combinationthereof. For example, a vocoder decoder 538 may be included in areceiver data processor 564. As another example, a vocoder encoder 536may be included in a transmission data processor 567.

The transcoder 510 may function to transcode messages and data betweentwo or more networks. The transcoder 510 may be configured to convertmessage and audio data from a first format (e.g., a digital format) to asecond format. To illustrate, the vocoder decoder 538 may decode encodedsignals having a first format and the vocoder encoder 536 may encode thedecoded signals into encoded signals having a second format.Additionally or alternatively, the transcoder 510 may be configured toperform data rate adaptation. For example, the transcoder 510 maydownconvert a data rate or upconvert the data rate without changing aformat the audio data. To illustrate, the transcoder 510 may downconvert64 kbit/s signals into 16 kbit/s signals.

The audio CODEC 508 may include the vocoder encoder 536 and the vocoderdecoder 538. The vocoder encoder 536 may include an encode selector, aspeech encoder, and a music encoder, as described with reference to FIG.4. The vocoder decoder 538 may include a decoder selector, a speechdecoder, and a music decoder.

The base station 500 may include a memory 532. The memory 532, such as acomputer-readable storage device, may include instructions. Theinstructions may include one or more instructions that are executable bythe processor 506, the transcoder 510, or a combination thereof, toperform the method 300 of FIG. 3A, the method 350 of FIG. 3B, or acombination thereof. The base station 500 may include multipletransmitters and receivers (e.g., transceivers), such as a firsttransceiver 552 and a second transceiver 554, coupled to an array ofantennas. The array of antennas may include a first antenna 542 and asecond antenna 544. The array of antennas may be configured towirelessly communicate with one or more wireless devices, such as thedevice 400 of FIG. 4. For example, the second antenna 544 may receive adata stream 514 (e.g., a bit stream) from a wireless device. The datastream 514 may include messages, data (e.g., encoded speech data), or acombination thereof.

The base station 500 may include a network connection 560, such asbackhaul connection. The network connection 560 may be configured tocommunicate with a core network or one or more base stations of thewireless communication network. For example, the base station 500 mayreceive a second data stream (e.g., messages or audio data) from a corenetwork via the network connection 560. The base station 500 may processthe second data stream to generate messages or audio data and providethe messages or the audio data to one or more wireless device via one ormore antennas of the array of antennas or to another base station viathe network connection 560. In a particular implementation, the networkconnection 560 may be a wide area network (WAN) connection, as anillustrative, non-limiting example. In some implementations, the corenetwork may include or correspond to a Public Switched Telephone Network(PSTN), a packet backbone network, or both.

The base station 500 may include a media gateway 570 that is coupled tothe network connection 560 and the processor 506. The media gateway 570may be configured to convert between media streams of differenttelecommunications technologies. For example, the media gateway 570 mayconvert between different transmission protocols, different codingschemes, or both. To illustrate, the media gateway 570 may convert fromPCM signals to Real-Time Transport Protocol (RTP) signals, as anillustrative, non-limiting example. The media gateway 570 may convertdata between packet switched networks (e.g., a Voice Over InternetProtocol (VoIP) network, an IP Multimedia Subsystem (IMS), a fourthgeneration (4G) wireless network, such as LTE, WiMax, and UMB, etc.),circuit switched networks (e.g., a PSTN), and hybrid networks (e.g., asecond generation (2G) wireless network, such as GSM, GPRS, and EDGE, athird generation (3G) wireless network, such as WCDMA, EV-DO, and HSPA,etc.).

Additionally, the media gateway 570 may include a transcoder, such asthe transcoder 510, and may be configured to transcode data when codecsare incompatible. For example, the media gateway 570 may transcodebetween an Adaptive Multi-Rate (AMR) codec and a G.711 codec, as anillustrative, non-limiting example. The media gateway 570 may include arouter and a plurality of physical interfaces. In some implementations,the media gateway 570 may also include a controller (not shown). In aparticular implementation, the media gateway controller may be externalto the media gateway 570, external to the base station 500, or both. Themedia gateway controller may control and coordinate operations ofmultiple media gateways. The media gateway 570 may receive controlsignals from the media gateway controller and may function to bridgebetween different transmission technologies and may add service toend-user capabilities and connections.

The base station 500 may include a demodulator 562 that is coupled tothe transceivers 552, 554, the receiver data processor 564, and theprocessor 506, and the receiver data processor 564 may be coupled to theprocessor 506. The demodulator 562 may be configured to demodulatemodulated signals received from the transceivers 552, 554 and to providedemodulated data to the receiver data processor 564. The receiver dataprocessor 564 may be configured to extract a message or audio data fromthe demodulated data and send the message or the audio data to theprocessor 506.

The base station 500 may include a transmission data processor 567 and atransmission multiple input-multiple output (MIMO) processor 568. Thetransmission data processor 567 may be coupled to the processor 506 andthe transmission MIMO processor 568. The transmission MIMO processor 568may be coupled to the transceivers 552, 554 and the processor 506. Insome implementations, the transmission MIMO processor 568 may be coupledto the media gateway 570. The transmission data processor 567 may beconfigured to receive the messages or the audio data from the processor506 and to code the messages or the audio data based on a coding scheme,such as CDMA or orthogonal frequency-division multiplexing (OFDM), as anillustrative, non-limiting examples. The transmission data processor 567may provide the coded data to the transmission MIMO processor 568.

The coded data may be multiplexed with other data, such as pilot data,using CDMA or OFDM techniques to generate multiplexed data. Themultiplexed data may then be modulated (i.e., symbol mapped) by thetransmission data processor 567 based on a particular modulation scheme(e.g., Binary phase-shift keying (“BPSK”), Quadrature phase-shift keying(“QSPK”), M-ary phase-shift keying (“M-PSK”), M-ary Quadrature amplitudemodulation (“M-QAM”), etc.) to generate modulation symbols. In aparticular implementation, the coded data and other data may bemodulated using different modulation schemes. The data rate, coding, andmodulation for each data stream may be determined by instructionsexecuted by processor 506.

The transmission MIMO processor 568 may be configured to receive themodulation symbols from the transmission data processor 567 and mayfurther process the modulation symbols and may perform beamforming onthe data. For example, the transmission MIMO processor 568 may applybeamforming weights to the modulation symbols. The beamforming weightsmay correspond to one or more antennas of the array of antennas fromwhich the modulation symbols are transmitted.

During operation, the second antenna 544 of the base station 500 mayreceive a data stream 514. The second transceiver 554 may receive thedata stream 514 from the second antenna 544 and may provide the datastream 514 to the demodulator 562. The demodulator 562 may demodulatemodulated signals of the data stream 514 and provide demodulated data tothe receiver data processor 564. The receiver data processor 564 mayextract audio data from the demodulated data and provide the extractedaudio data to the processor 506.

The processor 506 may provide the audio data to the transcoder 510 fortranscoding. The vocoder decoder 538 of the transcoder 510 may decodethe audio data from a first format into decoded audio data and thevocoder encoder 536 may encode the decoded audio data into a secondformat. In some implementations, the vocoder encoder 536 may encode theaudio data using a higher data rate (e.g., upconvert) or a lower datarate (e.g., downconvert) than received from the wireless device. Inother implementations the audio data may not be transcoded. Althoughtranscoding (e.g., decoding and encoding) is illustrated as beingperformed by a transcoder 510, the transcoding operations (e.g.,decoding and encoding) may be performed by multiple components of thebase station 500. For example, decoding may be performed by the receiverdata processor 564 and encoding may be performed by the transmissiondata processor 567. In other implementations, the processor 506 mayprovide the audio data to the media gateway 570 for conversion toanother transmission protocol, coding scheme, or both. The media gateway570 may provide the converted data to another base station or corenetwork via the network connection 560.

The vocoder decoder 538, the vocoder encoder 536, or both may receivethe parameter data and may identify the parameter data on aframe-by-frame basis. The vocoder decoder 538, the vocoder encoder 536,or both may classify, on a frame-by-frame basis, the synthesized signalbased on the parameter data. The synthesized signal may be classified asa speech signal, a non-speech signal, a music signal, a noisy speechsignal, a background noise signal, or a combination thereof. The vocoderdecoder 538, the vocoder encoder 536, or both may select a particulardecoder, encoder, or both based on the classification. Encoded audiodata generated at the vocoder encoder 536, such as transcoded data, maybe provided to the transmission data processor 567 or the networkconnection 560 via the processor 506.

The transcoded audio data from the transcoder 510 may be provided to thetransmission data processor 567 for coding according to a modulationscheme, such as OFDM, to generate the modulation symbols. Thetransmission data processor 567 may provide the modulation symbols tothe transmission MIMO processor 568 for further processing andbeamforming. The transmission MIMO processor 568 may apply beamformingweights and may provide the modulation symbols to one or more antennasof the array of antennas, such as the first antenna 542 via the firsttransceiver 552. Thus, the base station 500 may provide a transcodeddata stream 516, that corresponds to the data stream 514 received fromthe wireless device, to another wireless device. The transcoded datastream 516 may have a different encoding format, data rate, or both,than the data stream 514. In other implementations, the transcoded datastream 516 may be provided to the network connection 560 fortransmission to another base station or a core network.

The base station 500 may therefore include a computer-readable storagedevice (e.g., the memory 532) storing instructions that, when executedby a processor (e.g., the processor 506 or the transcoder 510), causethe processor to perform operations including decoding an encoded audiosignal to generate a synthesized signal. The operations may also includeclassifying the synthesized signal based on at least one parameterdetermined from the encoded audio signal.

The techniques described above may be compatible with Third GenerationPartnership Project (3GPP) networks. For example, quality of experienceacross particular metrics (e.g., speech quality, speech intelligibility,error resiliency, and call capacity) of AMR-WB in today's commercialVoLTE networks is: AMR-WB at 12.65 kbps operating over a unicast LTE PSchannel with 1% FER per mobile link (as specified for QCI=1) resultingin 2% total FER in mobile-to-mobile calls. About 90% of the cell areahas an end-to-end FER<=2%. This may be interpreted to mean that thereference “HD Voice” coverage is equivalent to 90% of the cell area. Inthe remaining 10% the AMR-WB codec speech quality starts to degrade atFERs above 2%.

The MCPTT service can be operated over three types of bearers dependingon the network topology that is most appropriate among those available.MCPTT can be operated over unicast channels in the same way theteleconferencing is performed in today's mobile networks using a centralconferencing server for duplicating and distributing media. Each of theLTE unicast channels may be a power-controlled channel that also usesretransmission schemes such as HARQ to provide a target BLER or packetloss rate to the VoIP frames transmitted over the channel. When usingAMR-WB in this topology, the coverage, error-resiliency, speech quality,speech intelligibility, and call capacity may be equivalent to that of“HD Voice.”

When multiple participants in a group are in a single cell the systemcan reduce the resources to support the users by having the users sharea common downlink MBMS bearer that is not power-controlled. There may beno dynamic feedback by which the eNB can decide to dynamically adjustits transmission resources to improve error performance or meet a targeterror rate. The use of retransmissions is “blind” in that theretransmissions are not sent based on dynamic feedback such asACK/NACKs. These retransmissions cannot be used to guarantee a certainlevel of performance or target error rate throughout the cell.Therefore, error rates on the MBMS bearer can vary considerablythroughout the cell, e.g., indoors, basements, elevators, stairwells, orthe edge of cell in an SC-PTM topology.

The topology for using an MBMS bearer may be configured as a Single-CellPoint-to-Multipoint (SC-PTM) bearer where adjacent cells do notnecessarily transmit the same group's content on the same MBMS bearer.In this topology the adjacent cells typically interfere with the MBMSbearer in the serving cell resulting in poorer coverage than the MBSFNtopology. The topology may also be part of a MBSFN, where all the cellsare broadcasting the same content on the same MBMS bearers, preventinginter-cell interference and allowing the users to combine thesetransmissions to improve coverage and reception. Unless the MCPTT groupis very large and spanning a large proportion of an operator's network,the system will most likely use SC-PTM in the cells serving the group asthis uses less of the overall network resources.

LTE-Direct communication is a broadcast mechanism (no physical layerfeedback) that defines two physical channels, control and data, forcommunication between two (or more) UEs. The resources used for directcommunication comprise of control and data resources. For in-networkoperation, a control resource pool is provided via RRC signaling whilefor off-network operation, the control resource pool is pre-configured.Further, two modes of resource allocation are supported: Mode 1(in-network) and Mode 2 (in-network and off-network). In Mode 2, thetransmitting UEs determine the resources to be used for control and datatransmission. UE transmits control to announce resources to be used forsubsequent data transmission. Receiving UEs monitor the controlresources to determine when to wake-up and listen for data transmission.

Enhanced Voice Services (EVS) is a new speech codec standard (part of3GPP Release 12) which offers a wide range of new features andimprovements for low delay real-time communication. Key advancementsinclude significantly improved quality for clean/noisy speech and musiccontent, higher compression efficiency and unprecedented errorresiliency to packet loss and delay jitter experienced in PS systems.

In general, EVS-WB codec offers quality significantly better than AMR-WBat a similar bit-rate and quality equivalent to AMR-WB at a lower bitrate. The EVS-SWB codec performance is significantly better than bothAMR-WB and corresponding bit rates of EVS-WB. For clean speech content,the lowest bit-rate of EVS-WB, namely 5.9 kbps, can offer qualitysignificantly better than AMR-WB at 8.85 kbps and equivalent to AMR-WBat 12.65 kbps. The subjective quality of EVS-WB coding starting at 9.6kbps is significantly better than the AMR-WB coding at its highest bitrate of 23.85 kbps. The super-wideband mode of EVS at 13.2 kbps achievestransparency to the direct source and offers quality significantlybetter than both 23.85 kbps of AMR-WB and 24.4 kbps of EVS-WB.

For noisy speech, EVS-WB at 9.6 kbps offers quality on par with AMR-WBat 12.65 kbps. This has also been shown across different languages/noisetypes and summarized in TR 26.952. However, none of the noisy speechtests included the presence of a front end noise suppression, which isexpected to establish the equivalence to AMR-WB 12.65 kbps quality at abit-rate lower than 9.6 kbps by providing a higher SNR at the input tothe coder. EVS-WB at 13.2 kbps offers quality on par with AMR-WB atapproximately twice the bit-rate with consistent progression insubjective quality with increasing bit-rates. The subjective quality ofEVS-SWB coding at 13.2 kbps is significantly better than that of AMR-WBat 23.85 kbps and EVS-WB at the same bit-rate. For mixed/music codingunder clean channel conditions, both the EVS-WB and SWB codec startingat 13.2 kbps achieves subjective quality that is significantly betterthan that of AMR-WB at any bit-rate. For North American English musicand mixed content, EVS-SWB coding performs significantly better thanEVS-WB at the same bit rate.

Most 3GPP networks are expected to be configured such that the FER isaround 1% for each link. While the 2% data point was not tested in thistest, comparisons are made between the EVS modes versus AMR-WB at thenearest data points namely, 0% (clean channel) and 3% FER. In general,the 13.2 kbps EVS-SWB clean speech performance under impaired channelwith channel aware mode enabled is significantly better than withoutchannel aware mode which in turn is significantly better than AMR-WB atits highest bit-rate of 23.85 kbps. For both languages, the quality ofEVS SWB 13.2 kbps channel aware and non-channel aware modes in cleanchannel are significantly better than AMR-WB at its highest bit-rate of23.85 kbps. For North American English, EVS SWB 13.2 kbps channel awaremode operating at around 6% FER delivers quality on par with that of thehighest bit-rate of AMR-WB (23.85 kbps) under no loss. The 13.2 kbps SWBnon-channel aware mode is able to achieve the quality equivalence toAMR-WB 23.85 clean channel when operating at around 3% FER. EVS 13.2kbps channel aware mode even at 10% FER delivers quality better thanAMR-WB 23.85 kbps at 3% FER, while the 13.2 kbps EVS SWB non-channelaware mode can operate at 8% FER but achieve quality equivalence toAMR-WB 23.85 kbps at 3% FER. For Danish, EVS SWB 13.2 kbps channel awaremode operating at around 3% FER delivers quality on par with that of thehighest bit-rate of AMR-WB (23.85 kbps) under no loss. EVS 13.2 kbpschannel aware mode even at 10% FER delivers quality equivalent to AMR-WB23.85 kbps at 3% FER, while the 13.2 kbps EVS SWB non-channel aware modecan operate at 6% FER to achieve quality equivalence to AMR-WB 23.85kbps at 3% FER.

In general, the 13.2 kbps EVS-SWB noisy speech performance underimpaired channel with channel aware mode enabled is significantly betterthan without channel aware mode which in turn is significantly betterthan AMR-WB at its highest bit-rate of 23.85 kbps. For North AmericanEnglish with car noise at 15 dB SNR, EVS 13.2 kbps channel aware modeoperating at 10% FER and the EVS SWB non-channel aware mode operating at6% FER can achieve quality equivalence to AMR-WB 23.85 kbps at 3% FER.For both languages, the 13.2 kbps EVS-WB clean speech performance underimpaired channel with and without channel aware mode enabled issignificantly better than AMR-WB at 15.85 kbps. For North AmericanEnglish, the 13.2 kbps EVS-WB clean speech performance under impairedchannel with channel aware mode enabled is significantly better thanwithout channel aware mode which in turn is significantly better thanAMR-WB at 15.85 kbps. The quality of EVS WB 13.2 kbps channel aware andnon-channel aware modes in clean channel are significantly better thanAMR-WB at 15.85 kbps. Specifically, it can be seen that for bothlanguages tested, the EVS 13.2 kbps channel aware mode operating at 10%FER can deliver quality on par with AMR-WB at 15.85 kbps at 3% FER. Inaddition, the 13.2 kbps non-channel aware mode can operate at 6% FER toachieve equivalence to AMR-WB 15.85 kbps at 3% FER.

Power controlled LTE unicast channels are typically configured tooperate at a target BLER of 1% per link. The EVS-SWB mode operating at13.2 kbps (channel aware and non channel aware) offers significantlybetter audio quality than EVS-WB at the same bit-rate. This applies to awide range of input signals which include clean speech, speech withbackground noise and mixed/music content. EVS offers significant voicequality improvement over AMR-WB (HD voice). The improved robustness tobackground noise and resiliency to errors are particularly relevant toMCPTT service and in general is expected to result in better or at leastequal speech intelligibility to “HD voice”.

Although retransmission schemes such as HARQ maybe used for tightcontrol of the target BLER, due to the power limited uplink, the celledge or deep indoors may still experience higher BLER (>1%). Under theseconditions the EVS WB and SWB channel aware mode may offer significantlybetter speech quality than AMR-WB at 12.65 kbps due to improved errorresiliency. It can be appreciated that the EVS WB channel aware mode at13.2 kbps can tolerate up to 8% FER and still deliver the same speechquality as AMR-WB 12.65 kbps operating at 2% FER which is center of thecell “HD Voice” speech quality. The ability to sustain the link whiletolerating higher path loss results in improved link budget/coverage.The EVS SWB channel aware mode at 13.2 kbps can tolerate even higher FER(up to 10%) for further extending coverage while maintaining HD Voicecenter of cell speech quality. The 13.2 kbps EVS WB and SWB non-channelaware modes can also operate at higher FER and deliver “HD Voice” centerof cell speech quality thereby resulting in improved coverage, albeitlower than that of the channel aware mode. The 13.2 kbps EVS modesutilize the same transport block size as AMR-WB 12.65 kbps. This resultsin the same cell site voice capacity as AMR-WB 12.65 kbps. If coverageis kept constant, the improved error resiliency can be utilized forcapacity gains by increasing the packet loss rate by not transmittingthe packet at the UE itself. The power at which each packet istransmitted would not be lowered but this mechanism can result in powersavings at the UE due to reduced ON time or reduced number oftransmissions (analogous to DTX or blanking but for active speech).Incorporating this into the scheduler can reduce the average number ofresource blocks required thereby freeing up resources either to add moreusers or for best effort traffic. The capacity gains may be directlyproportional to the maximum FER rate at which the EVS mode can stillmaintain HD voice center of cell voice quality. At 13.2 kbps the EVS SWBchannel aware mode would offer the highest capacity gains followed byEVS WB channel aware, EVS SWB and WB non-channel aware modes. Forexample, reducing ON time by 10% (i.e. blanking 10% of the active framevocoder packets) when operating in EVS SWB channel aware mode can resultin 18% capacity gains measured in terms of number of additional usersper cell site. The lower bit-rates of EVS, namely the 5.9 VBR and 7.2kbps WB modes, can offer significant cell site voice capacityimprovements due to utilizing smaller transport block sizes/resourceblocks.

AMR-WB is unable to meet the reference error resilience, speech quality,and speech intelligibility in all scenarios except for the 1% FER cases.EVS meets or exceeds these reference KPIs in most cases. In addition,both EVS wideband and superwideband modes offer notable improvement invoice quality over AMR-WB under various error conditions. In certaincases the improvement in P.OLQA scores by EVS over AMR-WB is 0.65, whileAMR-WB yield low P.OLQA scores around 2.84. This may translate intoscenarios where an end user may not understand certain portions ofspeech with AMR-WB under certain FER conditions in the MBMS coveragearea, while EVS would provide clear and meaningful speech. AMR-WB isunable to meet the reference “HD Voice” quality as it suffers from asignificant reduction in voice quality during channel errors. From acoverage perspective, this would be experienced as significantdegradation in voice quality in areas exhibiting above channelcharacteristics with errors.

Those of skill would further appreciate that the various illustrativelogical blocks, configurations, modules, circuits, and algorithm stepsdescribed in connection with the implementations disclosed herein may beimplemented as electronic hardware, computer software executed by aprocessor, or combinations of both. Various illustrative components,blocks, configurations, modules, circuits, and steps have been describedabove generally in terms of their functionality. Whether suchfunctionality is implemented as hardware or processor executableinstructions depends upon the particular application and designconstraints imposed on the overall system. Skilled artisans mayimplement the described functionality in varying ways for eachparticular application, such implementation decisions are not to beinterpreted as causing a departure from the scope of the presentdisclosure.

The steps of a method or algorithm described in connection with theimplementations disclosed herein may be embodied directly in hardware,in a software module executed by a processor, or in a combination of thetwo. A software module may reside in random access memory (RAM), flashmemory, read-only memory (ROM), programmable read-only memory (PROM),erasable programmable read-only memory (EPROM), electrically erasableprogrammable read-only memory (EEPROM), registers, hard disk, aremovable disk, a compact disc read-only memory (CD-ROM), or any otherform of non-transient storage medium known in the art. An exemplarystorage medium is coupled to the processor such that the processor mayread information from, and write information to, the storage medium. Inthe alternative, the storage medium may be integral to the processor.The processor and the storage medium may reside in an ASIC. The ASIC mayreside in a computing device or a user terminal. In the alternative, theprocessor and the storage medium may reside as discrete components in acomputing device or user terminal.

The previous description of the disclosed implementations is provided toenable a person skilled in the art to make or use the disclosedimplementations. Various modifications to these implementations will bereadily apparent to those skilled in the art, and the principles definedherein may be applied to other implementations without departing fromthe scope of the disclosure. Thus, the present disclosure is notintended to be limited to the implementations shown herein and is to beaccorded the widest scope possible consistent with the principles andnovel features as defined by the following claims.

What is claimed is:
 1. A method of controlling frame transmissions, themethod comprising: determining, at a first device, a frame erasure ratefor a communication session between the first device and at least asecond device; comparing the frame erasure rate to an erasure threshold;and discarding an active speech frame if the frame erasure ratesatisfies the erasure threshold.
 2. The method of claim 1, whereindiscarding the active speech frame comprises bypassing transmission ofthe active speech frame.
 3. The method of claim 1, wherein the frameerasure rate satisfies the erasure threshold if the frame erasure rateis lower than the erasure threshold.
 4. The method of claim 1, furthercomprising receiving a signal indicating the erasure threshold prior tocomparing the frame erasure rate to the erasure threshold, the signalreceived from a network device.
 5. The method of claim 1, wherein thefirst device stores data indicating the erasure threshold.
 6. The methodof claim 1, further comprising: in response to determining that frameerasure rate satisfies the erasure threshold, determining if aparticular active speech frame includes a partial copy of a previousframe; and determining whether the previous frame was transmitted to thesecond device in response to determining that the particular activespeech frame includes the partial copy of the previous frame.
 7. Themethod of claim 6, further comprising transmitting the particular activespeech frame to the second device if the previous frame was nottransmitted to the second device.
 8. The method of claim 6, furthercomprising, if the previous frame was transmitted to the second device,comparing a first criticality value of the particular active speechframe to a first threshold and comparing a second criticality value ofthe previous frame to a second threshold.
 9. The method of claim 8,further comprising: transmitting the particular active speech frame tothe second device if the first criticality value is not less than thefirst threshold or if the second criticality value is not less than thesecond threshold; and discarding the particular active speech frame ifthe first criticality value is less than the first threshold and thesecond criticality value is less than the second threshold.
 10. Themethod of claim 6, further comprising determining whether a firstcriticality value of the particular active speech frame is less than athird threshold in response to determining that the particular activespeech frame does not include the partial copy of the previous frame.11. The method of claim 10, further comprising transmitting theparticular active speech frame to the second device if a firstcriticality value of the particular active speech frame is not less thanthe third threshold.
 12. The method of claim 10, further comprisingdiscarding the particular active speech frame if a first criticalityvalue of the particular active speech frame is less than the thirdthreshold.
 13. The method of claim 1, wherein discarding the activespeech frame is performed within a device that comprises a mobiledevice.
 14. The method of claim 1, wherein discarding the active speechframe is performed within a device that comprises a base station. 15.The method of claim 1, further comprising receiving a signal indicatingthe frame erasure rate, the signal received from the second device. 16.The method of claim 1, wherein discarding the active speech framereduces an amount of power consumption at the first device.
 17. Themethod of claim 1, wherein discarding the active speech frame increasesnetwork capacity.
 18. An apparatus comprising: a rate monitor configuredto determine a frame erasure rate for a communication session between afirst device and at least a second device; comparison circuitryconfigured to compare the frame erasure rate to an erasure threshold;and active frame discard circuitry configured to discard an activespeech frame if the frame erasure rate satisfies the erasure threshold.19. The apparatus of claim 18, further comprising a transmitterconfigured to bypass transmission of the active speech frame to thesecond device.
 20. The apparatus of claim 18, wherein the frame erasurerate satisfies the erasure threshold if the frame erasure rate is lowerthan the erasure threshold.
 21. The apparatus of claim 18, furthercomprising a receiver configured to receive a signal indicating theerasure threshold prior to comparing the frame erasure rate to theerasure threshold, the signal received from a network device.
 22. Theapparatus of claim 18, further comprising a speech vocoder configured todetermine the erasure threshold based on an algorithm.
 23. The apparatusof claim 18, further comprising a speech vocoder configured to: inresponse to determining that frame erasure rate satisfies the erasurethreshold, determine if a particular active speech frame includes apartial copy of a previous frame; and determine whether the previousframe was transmitted to the second device in response to determiningthat the particular active speech frame includes the partial copy of theprevious frame.
 24. The apparatus of claim 23, further comprising atransmitter configured to transmit the particular active speech frame tothe second device if the previous frame was not transmitted to thesecond device.
 25. The apparatus of claim 23, wherein, if the previousframe was transmitted to the second device, the comparison circuitry isfurther configured to: compare a first criticality value of theparticular active speech frame to a first threshold; and compare asecond criticality value of the previous frame to a second threshold,and wherein the speech vocoder is further configured to: initiatetransmission of the particular active speech frame to the second deviceif the first criticality value is not less than the first threshold orif the second criticality value is not less than the second threshold;and discard the particular active speech frame if the first criticalityvalue is less than the first threshold and the second criticality valueis less than the second threshold.
 26. The apparatus of claim 18,further comprising: an antenna; and a receiver coupled to the antennaand configured to receive an encoded audio signal.
 27. The apparatus ofclaim 26, wherein the antenna, the receiver, the rate monitor, thecomparison circuitry, and the active frame discard circuitry areintegrated into a mobile device or a base station.
 28. A non-transitorycomputer-readable medium comprising instructions for controlling frametransmissions, the instructions, when executed by a processor, cause theprocessor to perform operations comprising: determining, at a firstdevice, a frame erasure rate for a communication session between thefirst device and at least a second device; comparing the frame erasurerate to an erasure threshold; and discarding an active speech frame ifthe frame erasure rate satisfies the erasure threshold.
 29. Thenon-transitory computer-readable medium of claim 28, wherein discardingthe active speech frame comprises bypassing transmission of the activespeech frame.
 30. An apparatus comprising: means for determining a frameerasure rate for a communication session between a first device and atleast a second device; means for comparing the frame erasure rate to anerasure threshold; and means for discarding an active speech frame ifthe frame erasure rate satisfies the erasure threshold.
 31. Theapparatus of claim 30, further comprising means for bypassingtransmission of the active speech frame.
 32. A method of controlling ablock error rate for a communication channel, the method comprising:determining, at a particular device, that a communication sessionbetween a first device and a second device supports an Enhanced VoiceServices (EVS) coder/decoder (CODEC), the first device and the seconddevice communicating via the communication channel; and increasing theblock error rate for the communication channel in response todetermining that the communication session supports the EVS CODEC. 33.The method of claim 32, wherein increasing the block error rate for thecommunication channel increases link budget gains for the communicationsession.
 34. The method of claim 32, wherein increasing the block errorrate for the communication channel increases an amount of coverage forthe communication session.
 35. The method of claim 32, wherein the EVSCODEC supports EVS channel aware mode.
 36. The method of claim 32,wherein the EVS CODEC supports 13.2 kbps non-channel aware mode.
 37. Themethod of claim 32, wherein the particular device comprises a networkdevice, the first device, or the second device.